[asterisk-users] sip server
Hi, Can i use asterisk as sip server for manage call Transmission between gateways Best Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip server
Yes CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mohamed daif Sent: Monday, June 28, 2010 2:49 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip server Hi, Can i use asterisk as sip server for manage call Transmission between gateways Best Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip server
hi i want to use asterisk as a sip server without installing any hardware in this machine the question is how can i configure the external getaways with asterisk how can i configure the costumer who is i provide calls to hem what is the billing software can i use to calculate the the calls and manage the rate -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip server
On Mon, Jun 28, 2010 at 4:02 PM, mohamed daif mohamed.d...@gmail.com wrote: i want to use asterisk as a sip server without installing any hardware in this machine the question is how can i configure the external getaways with asterisk how can i configure the costumer who is i provide calls to hem what is the billing software can i use to calculate the the calls and manage the rate Time to do some reading: http://astbook.asteriskdocs.org/ -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP server behind NAT
Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The Asterisk server connects to a remote VoIP provider via SIP. The extensions.conf and sip.conf follow below. I contacted the provider to see if there was a specific problem, and I was advised that my asterisk was not using standard ports for the transit of voice (I assume they mean RTP on UDP). They're telling me that, normally, the port 5004 should be used, or ports above 1. However for one of my outbound calls, they see me using port 3030. Could someone advise me on the steps to follow, or to documentation on this issue? Does this sound like a NAT issue? All the best, Guillaume Yziquel. Here's the beginning of the sip.conf file: [general] context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp' ;realm=mydomain.tld ; Realm for digest authentication bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;domain=mydomain.tld; Set default domain for this host ;domain=mydomain.tld,mydomain-incoming ;domain=1.2.3.4 ; Add IP address as local domain ;allowexternalinvites=no; Disable INVITE and REFER to non-local domains ;autodomain=yes ; Turn this on to have Asterisk add local host ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ;tos=184; Set IP QoS to either a keyword or numeric val tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpiry=3600 ; Max length of incoming registration we allow defaultexpiry=120 ; Default length of incoming/outgoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;checkmwi=10; Default time between mailbox checks for peers ;vmexten=voicemail ; dialplan extension to reach mailbox sets the ;videosupport=yes ; Turn on support for SIP video ;recordhistory=yes ; Record SIP history by default disallow=all; First disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=gsm ; musicclass=default ; Sets the default music on hold class for all SIP calls language=en ; Default language setting for all users/peers relaxdtmf=yes ; Relax dtmf handling rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity trustrpid = no ; If Remote-Party-ID should be trusted sendrpid = yes ; If Remote-Party-ID should be sent progressinband=no ; If we should generate in-band ringing always useragent=My Asterisk ; Allows you to change the user agent string promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ;usereqphone = no ; If yes, ;user=phone is added to uri that contains dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 ;compactheaders = yes ; send compact sip headers. ;sipdebug = yes ; Turn on SIP debugging by default, from ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests ;notifyringing = yes; Notify subscriptions on RINGING state ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected, ;regcontext=sipregistrations ;registertimeout=20 ; retry registration calls every 20 seconds (default) ;registerattempts=10; Number of registration attempts before we give up callevents=no ; generate manager events when sip ua performs events (e.g. hold) externip=The_IP_of_my_router; Address that we're going to put in outbound SIP messages ;externhost=foo.dyndns.net ; Alternatively you can specify an ;externrefresh=10 ; How often to refresh externhost if localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation localnet=169.254.0.0/255.255.0.0 ;Zero conf local network nat=yes ; Global NAT
[asterisk-users] SIP Server
Hi Dear Users, I am new to Asterisk and had a query which is probably primitive. I wanted to know whether I can use the Digium Hardware and receive and establish connection to a host SIP Server which is totally a different platform. Let me explain - Usually there is a E1-VoIP gateway (independent Hardware) connecting to a Server/Client via LAN. In my case, SIP Server. Now what I want is that Digium PCI Hardware and the SIP Server will be the same PC and I Want the PCI Hardware to act as the gateway. Therefore my question in particular is: That is can I configure the device to talk to the Server in SIP protocol directly?-- Imran M Yousuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Server
Yes, you just need to setup asterisk with Digium board on the same server of your sipserver, and then you must establish a trunk between your sip server and asterisk. Then you must route calls using asterisk dialplan as well as your sip server dialplan. Be aware that if you have both on same server you must change SIP port in one of them. On 10/30/06, Imran M Yousuf [EMAIL PROTECTED] wrote: Hi Dear Users, I am new to Asterisk and had a query which is probably primitive. I wanted to know whether I can use the Digium Hardware and receive and establish connection to a host SIP Server which is totally a different platform. Let me explain - Usually there is a E1-VoIP gateway (independent Hardware) connecting to a Server/Client via LAN. In my case, SIP Server. Now what I want is that Digium PCI Hardware and the SIP Server will be the same PC and I Want the PCI Hardware to act as the gateway. Therefore my question in particular is: That is can I configure the device to talk to the Server in SIP protocol directly? -- Imran M Yousuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Server question / recommendations
On Thursday 16 December 2004 01:09, Shahed wrote: Hello All, I am new to *, and this is my first post on the user list. I have had success with making / receiving calls to a SIP hardware Phone and the Console Channel Driver. Can anyone please suggest what would be a good SIP server to use, or is there a way in which I can use asterisk itself as a SIP server for my phone and make calls to it using the console ?? Yes, Asterisk is a SIP server - see /etc/asterisk/sip.conf Antony. -- Wanted: telepath. You know where to apply. Please reply to the list; please don't CC me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Server question / recommendations
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shahed Sent: December 15, 2004 8:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Server question / recommendations Hello All, I am new to *, and this is my first post on the user list. I have had success with making / receiving calls to a SIP hardware Phone and the Console Channel Driver. My SIP phone *requires* that I register with a SIP server. For this, I used the NIST sip presence server (a version that I downloaded almost a year ago). I have a problem with getting asterisk to register properly, (using domain names) etc, but I did get it to work. Can anyone please suggest what would be a good SIP server to use, or is there a way in which I can use asterisk itself as a SIP server for my phone and make calls to it using the console ?? Asterisk will do well if you just want to hook a few SIP phones up to it. For a really big SIP environment, you'll want to look at SER (Sip Express Router). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Server question / recommendations
Hello All, I am new to *, and this is my first post on the user list. I have had success with making / receiving calls to a SIP hardware Phone and the Console Channel Driver. My SIP phone *requires* that I register with a SIP server. For this, I used the NIST sip presence server (a version that I downloaded almost a year ago). I have a problem with getting asterisk to register properly, (using domain names) etc, but I did get it to work. Can anyone please suggest what would be a good SIP server to use, or is there a way in which I can use asterisk itself as a SIP server for my phone and make calls to it using the console ?? Thanks Shahed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Server
Hello: I am doing research on designing an architecture for a residential-based VoIP service. Meaning, the platform should ideally able to handle lots of SIP sessions at the same time. I understand Asterisk do come with SIP Proxy and Outbound Proxy, but it is not scalable...and I heard about SER is a better choice. And what database would you recommend. is MySQL robust and scalable enough? I appreciate any comments. Thanks, Danny ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
